Displaying 20 results from an estimated 2000 matches similar to: "One way audio when dialing multiple registrations"
2010 Jul 20
0
asterisk-users Digest, Vol 72, Issue 49
sorry for typo mistake in my last post. as from my orignal post two
registration of the same user are as follows
SIP/XYZ at 119.68.0.90:5060
SIP/XYZ at 202.16.34.10:5678
so dial command with unique-id i want to use will be
Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT)
and not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 24
Yes this scenario works on my 2 systems which are at LAN. I made one system
as server (192.168.0.20) and registered from other system... it is fine but
now there is a different scene.
actually there is a registered user named abc at system1 (192.168.0.20)
having context [payasyougo] which is used to do outbound calls. we want to
use this user's context and account so that when we register
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 23
Thanks Vardan,
I will like to know if this scenario can work when peer is not having fixed
ip and we use
host = nasir.server.com
?
also I have set insecure=invite,port
what if i use
insecure=no
thanks again.
Message: 24
Date: Tue, 11 May 2010 10:52:14 +0500
From: Vardan <hvardan71 at gmail.com>
Subject: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To:
2010 May 10
1
Dialing a SIP Peer without using register strin
Hi,
I am new to this list and this is first time i m posting here. please help
me out
currently I am working on dialing a sip peer on an asterisk server from 2nd
asterisk server. scenario is like this.
on my system i am using this peer in sip.conf.
[abc]
type=peer
username=abc
secret=mysecret
host=192.168.0.20
context=default
dtmfmode=rfc2833
;restrictcid=no
canreinvite=yes
2010 May 12
2
asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
2010 Aug 03
2
RTP stream not passing through router with port forwarding
Hi,
I am trying to dial a registered user via his IP:Port mechanism, but problem
is that the audio data is not reaching to dialed user. here is the scenario.
caller and callee both are registered at asterisk server. asterisk server is
on public ip so no port forwarding and natting necessary there. however
caller and callee both are behind router and there is port forwarding
enabled and nat=yes,
2010 Jul 28
2
Nat issue one way audio on IP dial
hi there,
i have posted earlier on the list but got no satisfying answer. the problem
is not big.
I have asterisk server directly connected with internet (79.80.x.x) and
clients are behind router. clients/users are registered with asterisk and
are using sipura and xlite softphone.
Now problem is that when a user calls other by dialing his IP:Port (sip
uri), call is connected fine and he can
2010 Jul 16
0
asterisk-users Digest, Vol 72, Issue 39
yes, actually this scenario is on remote servers. like
SIP/XYZ at 119.18.230.20:5060
SIP/XYZ at 202.68.0.90:5678
audio is ok when dialing without using ip & port as below
SIP/XYZ
but when i dial using below dialstring
SIP/XYZ at 202.68.0.90:5678
or
SIP/XYZ at 119.18.230.20:5060
then the problem arises
hope you got the idea..
Nasir
2010 Jul 29
4
How to extract channel-id of a user or peer
Hi,
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works correctly when user and asterisk are on
Local LAN. But my asterisk server is on public ip and users are behind nat,
and so this method says unknow host when used on public asterisk server.
I also tried built-in variable ${CHANNEL}, but this returns the channel-id
of the calling channel.
2010 Jul 22
0
SIP URI Dial has one way audio
Hi,
I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ
as target user which is registered.
Asterisk server IP: 70.118.x.x
calling user IP: 117.58.x.x
called user IP: 117.58.x.x:5062
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,30,rtT)
and during conversation audio was OK in both ways. Then I dialed the
registered user via
2018 Oct 04
3
CURL to post application/json
We tried to use the CURL fn to POST json, but it's sent as form data and
there seems no support for changing the Content-Type header. We switched to
invoking curl in the shell.
All the documentation I could find says there is just one parameter for the
url and an optional second for POST body. Is there an undocumented way to
set Content-Type?
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An HTML
2007 Jun 02
3
Dynamically adding Context in dialplan?
Hi everybody,
>From asterisk CLI we can add extensions in dial-plan dynamically using
"dialplan add extension" command.
but how we can dynamically create a context in dialplan. is that
possible?
Nasir Iqbal
2011 Mar 05
3
R Statistical Package Installation
Dear R-project team,
I have tried but could not install the R statistical package (http://cran.ms.unimelb.edu.au/ ) even after the help of my institute's IT personnel. The setup file could not be downloaded. The latest file R-2.12.2.tar.gz<http://cran.ms.unimelb.edu.au/src/base/R-2/R-2.12.2.tar.gz> does not start installation wizard. Kindly extend the technical support.
Best regards.
2018 Oct 03
2
WebRTC as Softphone substitute ?
@Olivior
I agree that seting up WebRTC is hard, however when done it is smooth to
use. For replication you can build RPMs with working configurations.
Regarding stability, it is not being used widly, so can't say it is mature.
However we have no complain so far regarding audio or connectivity.
sometime we provide support for "allow media / mic" type issues, but you
know it is
2003 Sep 15
2
Unable to access the mailbox or folders !!
Hi all,
I have installed dovecot on redhat linux with
ldap backend. I can login using ldap account in to my
webmail (squirrelmail) .But when I login to the
webmail , I cant see any inbox or anything . Just some
error messages like this ,
ERROR:
ERROR : Connection dropped by imap-server.
Query: LIST "" "Sent"
ERROR : Could not complete request.
Query: SELECT
2018 Sep 29
2
WebRTC as Softphone substitute ?
Hi Olivior,
We have recently worked on a WebRTC based agent panel. As based on my
experience I think that WebRTC based phones are far better and cheaper then
those soft / sip phone. the big plus is that they are easy to customize and
developer can use the power of browser and web to build / offer features
which are not possible with regular phones.
Regarding your concern about BLF or call
2005 Jun 09
1
IAX2 Max Retries dropped calls Firefly
Hi
We've recently set up and are using with success 1.0.7 using a Junghanns
quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works
very well, however we're getting cases where sometimes the call just drops.
>From setting more verbose modes we get a log which is shown below. The problem
seems to be the maxretries message which comes from chan_iax2. We are using
2007 Aug 05
2
FETCH for mailbox XXX UID XXX got too little data.
Hi Timo,
I am receiving the above error in dovecot's logs, here's how it looks
like:
dovecot: Aug 05 19:52:19 Info: imap-login: Login: user=<XXXXXXXXXXXX>,
method=PLAIN, rip=127.0.0.1, lip=127.0.0.1, secured
dovecot: Aug 05 19:53:36 Error: IMAP(XXXXXXXXXXXX): FETCH for mailbox Bulk Mails UID 32
got too little data: 9199 vs 39532
dovecot: Aug 05 19:53:36 Info: IMAP(XXXXXXXXXXXX):
2007 May 31
5
Auto Dial Problem
Hi All,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
My sample call file :
hannel: local/0124787924@outbound-reminder
MaxRetries: 5
RetryTime: 300
WaitTime: 40
Account: Reminder
context: remindem
extension: s
priority: 1
Set: MSG=0135.20070601.0124787924
Set:
2010 Aug 06
4
How do I install speex for asterisk?
Hi,
I have followed steps which were mentioned on forum and given below. Still
couldn't get speex working. On test calls getting error "chan_sip.c:
sip_call: No audio format found to offer."
# yum install speex
# yum install speex-devel
# cd /usr/src/asterisk
# make clean
# make
# service asterisk stop
# make install
# service asterisk start
Also, it is not